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---
license: mit
language:
- en
- ja
- nl
- fr
- de
- it
- pl
- pt
- es
- ko
- zh
tags:
- speech
- audio
- tokenizer
datasets:
- sarulab-speech/mls_sidon
- mythicinfinity/Libriheavy-HQ
- nvidia/hifitts-2
- amphion/Emilia-Dataset
pipeline_tag: audio-to-audio
---

# MioCodec-25Hz-24kHz: Lightweight Neural Audio Codec for Efficient Spoken Language Modeling

[![GitHub](https://img.shields.io/badge/Code-GitHub-black)](https://github.com/Aratako/MioCodec)

**MioCodec-25Hz-24kHz** is a lightweight and fast neural audio codec designed for efficient spoken language modeling. Based on the [Kanade-Tokenizer](https://github.com/frothywater/kanade-tokenizer) implementation, this model features an integrated wave decoder (iSTFTHead) that directly synthesizes waveforms without requiring an external vocoder.

For higher audio fidelity at 44.1 kHz, see [MioCodec-25Hz-44.1kHz](https://huggingface.co/Aratako/MioCodec-25Hz-44.1kHz).

## ๐ŸŒŸ Overview

MioCodec decomposes speech into two distinct components:

1.  **Content Tokens:** Discrete representations that primarily capture linguistic information and phonetic content ("what" is being said) at a low frame rate (25 Hz).
2.  **Global Embeddings:** A continuous vector representing broad acoustic characteristics ("how")โ€”including speaker identity, recording environment, and microphone traits.

By disentangling these elements, MioCodec is ideal for **Spoken Language Modeling**.

### Key features

* **Lightweight & Fast:** Integrated wave decoder (iSTFTHead) enables direct waveform synthesis without an external vocoder.
* **Ultra-Low Bitrate:** Achieves high-fidelity reconstruction at only **341 bps**.
* **End-to-End Design:** Single model architecture from audio input to waveform output.

## ๐Ÿ“Š Model Comparison

| Model | Token Rate | Vocab Size | Bit Rate | Sample Rate | SSL Encoder | Vocoder | Parameters | Highlights |
| :--- | :---: | :---: | :---: | :---: | :---: | :---: | :---: | :--- |
| **MioCodec-25Hz-24kHz** | **25 Hz** | **12,800** | **341 bps** | **24 kHz** | **WavLM-base+** | **- (iSTFTHead)** | **132M** | **Lightweight, fast inference** |
| MioCodec-25Hz-44.1kHz | 25 Hz | 12,800 | 341 bps | 44.1 kHz | WavLM-base+ | [MioVocoder](https://huggingface.co/Aratako/MioVocoder) (Jointly Tuned) | 118M (w/o vocoder) | High-quality, high sample rate |
| kanade-25hz | 25 Hz | 12,800 | 341 bps | 24 kHz | WavLM-base+ | Vocos 24kHz | 118M (w/o vocoder) | Original 25Hz model |
| kanade-12.5hz | 12.5 Hz | 12,800 | 171 bps | 24 kHz | WavLM-base+ | Vocos 24kHz | 120M (w/o vocoder) | Original 12.5Hz model |

## ๐Ÿš€ Quick Start

### Installation

```bash
# Install via pip
pip install git+https://github.com/Aratako/MioCodec

# Or using uv
uv add git+https://github.com/Aratako/MioCodec

```

### Basic Inference

Basic usage for encoding and decoding audio:

```python
from miocodec import MioCodecModel, load_audio
import soundfile as sf

# 1. Load model
model = MioCodecModel.from_pretrained("Aratako/MioCodec-25Hz-24kHz").eval().cuda()

# 2. Load audio
waveform = load_audio("input.wav", sample_rate=model.config.sample_rate).cuda()

# 3. Encode Audio
features = model.encode(waveform)

# 4. Decode to Waveform (directly, no vocoder needed)
resynth = model.decode(
    content_token_indices=features.content_token_indices,
    global_embedding=features.global_embedding,
)

# 5. Save
sf.write("output.wav", resynth.cpu().numpy(), model.config.sample_rate)
```

### Voice Conversion (Zero-shot)

MioCodec allows you to swap speaker identities by combining the content tokens of a source with the global embedding of a reference.

```python
source = load_audio("source_content.wav", sample_rate=model.config.sample_rate).cuda()
reference = load_audio("target_speaker.wav", sample_rate=model.config.sample_rate).cuda()

# Perform conversion
vc_wave = model.voice_conversion(source, reference)
sf.write("converted.wav", vc_wave.cpu().numpy(), model.config.sample_rate)
```

## ๐Ÿ—๏ธ Training Methodology

MioCodec-25Hz-24kHz was trained in two phases with an integrated wave decoder that directly synthesizes waveforms via iSTFT.

### Phase 1: Feature Alignment

The model is trained to minimize both **Multi-Resolution Mel-spectrogram loss** and **SSL feature reconstruction loss** (using WavLM-base+). The wave decoder directly generates waveforms, and losses are computed on the reconstructed audio.

* **Multi-Resolution Mel Spectrogram Loss:** Using window lengths of `[32, 64, 128, 256, 512, 1024, 2048]`.
* **SSL Feature Reconstruction Loss:** Using WavLM-base+ features.

### Phase 2: Adversarial Refinement

Building upon Phase 1, adversarial training is introduced to improve perceptual quality. The training objectives include:

* **Multi-Resolution Mel Spectrogram Loss:** Using window lengths of `[32, 64, 128, 256, 512, 1024, 2048]`.
* **SSL Feature Reconstruction Loss:** Using WavLM-base+ features.
* **Multi-Period Discriminator (MPD):** Using periods of `[2, 3, 5, 7, 11, 17, 23]`.
* **Multi-Scale STFT Discriminator (MS-STFTD):** Using FFT sizes of `[118, 190, 310, 502, 814, 1314, 2128, 3444]`.
* **RMS Loss:** To stabilize energy and volume.

## ๐Ÿ“š Training Data

The training datasets are listed below:

| Language | Approx. Hours | Dataset |
| :--- | :--- | :--- |
| **Japanese** | ~22,500h | Various public HF datasets |
| **English** | ~500h | [Libriheavy-HQ](https://huggingface.co/datasets/mythicinfinity/Libriheavy-HQ) |
| **English** | ~4,000h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **English** | ~9,000h | [HiFiTTS-2](https://huggingface.co/datasets/nvidia/hifitts-2) |
| **English** | ~27,000h | [Emilia-YODAS](https://huggingface.co/datasets/amphion/Emilia-Dataset) |
| **German** | ~1,950h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **German** | ~5,600h | [Emilia-YODAS](https://huggingface.co/datasets/amphion/Emilia-Dataset) |
| **Dutch** | ~1,550h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **French** | ~1,050h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **French** | ~7,400h | [Emilia-YODAS](https://huggingface.co/datasets/amphion/Emilia-Dataset) |
| **Spanish** | ~900h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **Italian** | ~240h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **Portuguese** | ~160h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **Polish** | ~100h | [MLS-Sidon](https://huggingface.co/datasets/sarulab-speech/mls_sidon) |
| **Korean** | ~7,300h | [Emilia-YODAS](https://huggingface.co/datasets/amphion/Emilia-Dataset) |
| **Chinese** | ~300h | [Emilia-YODAS](https://huggingface.co/datasets/amphion/Emilia-Dataset) |

## ๐Ÿ“œ Acknowledgements

* **Codec Architecture:** Based on the brilliant work of [kanade-tokenizer](https://github.com/frothywater/kanade-tokenizer).
* **Decoder Design:** Inspired by [XCodec2](https://github.com/zhenye234/X-Codec-2.0).
* **Training Techniques:** Training objectives were inspired by [XCodec2](https://github.com/zhenye234/X-Codec-2.0) and [Inworld TTS-1](https://arxiv.org/html/2507.21138v1).

## ๐Ÿ–Š๏ธ Citation

```bibtex
@misc{miocodec-25hz-24khz,
  author = {Chihiro Arata},
  title = {MioCodec: High-Fidelity Neural Audio Codec for Efficient Spoken Language Modeling},
  year = {2026},
  publisher = {Hugging Face},
  journal = {Hugging Face repository},
  howpublished = {\url{https://huggingface.co/Aratako/MioCodec-25Hz-24kHz}}
}
```